Issues 20000 - 20999

[..]
ASTERISK-20003: Asterisk voice broadcast with SS7
ASTERISK-20004: DNS manager - multiple queries
ASTERISK-20006: Fix NULL pointer segfault in ast_sockaddr_parse()
ASTERISK-20007: GotoIf() documentation updates to be more clear that [[context,]extension,]priority is valid
ASTERISK-20008: outboundproxy ignored after when sending invite after 407
ASTERISK-20010: Bridge CDR fails to be created when, in a Local channel bridging two outbound channels that is optimized away, the CDR userfield setting is set on both outbound channels
ASTERISK-20012: Problems with parking calls in 1.8.14.0-rc1 that were not present in 1.8.13.0
ASTERISK-20013: Asterisk receiving RFC2833 DTMF with out of order sequence numbers from a Centile IntaSwitch Server results in DTMF digits being dropped
ASTERISK-20014: Voicemail giving incorrect length and being dropped
ASTERISK-20015: Device handling issues in skinny
ASTERISK-20016: Caller ID for Saudi Arabia
ASTERISK-20017: [patch] Multiple bugs in the handling of dynamic DUNDi peers
ASTERISK-20018: Asterisk 1.8.14.0 Blockers
ASTERISK-20019: Asterisk 10.6.0 Blockers
ASTERISK-20020: Write call parking test plan
ASTERISK-20022: CLONE - CLI hang and unresponsive when issuing "show channels" or "core show channels"
ASTERISK-20023: double uri_escaping of contact in outbound invite
ASTERISK-20024: India CallerID DTMF apparent in audio captured from TDM410 may not be reliably detected within Asterisk
ASTERISK-20028: Join message played twice to Bridge channel
ASTERISK-20029: When running a macro or subroutine callback via CCBS, dialplan execution continues unexpectedly
ASTERISK-20030: Switching from wifi to 3g on a mobile device connected to Asterisk via active SIP channel results in no audio
ASTERISK-20031: Asterisk 1.6.2.23 Deadlock
ASTERISK-20032: Call to timerfd_gettime() error: Bad file descriptor
ASTERISK-20036: Different behavior in 1.8.12.0 and 1.8.13.0 than 1.8.9.X
ASTERISK-20037: func_odbc timeout quite long when connection drops during query - possibly need to expose query timeout attribute to make it configurable
ASTERISK-20038: Fix locking usage in ExternalIVR
ASTERISK-20039: DTMF meta-digit W missing
ASTERISK-20040: Asterisk crashes when a guest call uses directmedia
ASTERISK-20041: Recorded file with MixMonitor ignore value defined with PITCH_SHIFT function
ASTERISK-20042: Voice Mail with IMAP storage wrong sequence number.
ASTERISK-20043: Queues with linear ring strategy
ASTERISK-20044: Crash in sip_setoption - chan_sip.c
ASTERISK-20046: CDR data in userfield for called channel is duplicated when set on calling and called channels
ASTERISK-20047: 1.8.13.0 Unable to get own IP address, SIP disabled
ASTERISK-20048: DTMF not working for cell phones with Asterisk IVR
ASTERISK-20049: Double PROGRESS message when overlap dialing is enabled
ASTERISK-20050: intermittent one way audio
ASTERISK-20051: Research addition of DTLS-SRTP support
ASTERISK-20052: Security Vulnerability: remote crash vulnerability in app_voicemail
ASTERISK-20054: Create Bridging Tests
ASTERISK-20058: Unable to receive inbound CallerID from analog PSTN circuit using Xorcom XR0030 FXO
ASTERISK-20059: [patch] Set variable with result code on a SIP Transfer
ASTERISK-20060: fix suggested for a misleading warning when getting a 408
ASTERISK-20061: PHP Agi Socket Server stopped working - socket_read spamming Resource temporarily unavailable
ASTERISK-20064: Can MWI frequency be reduced? pollmailboxes=yes results in double MWI
ASTERISK-20066: CDR(dst) is set to s when destination is BUSY on AMI Originate
ASTERISK-20068: CDR(dst) is set to s when destination is BUSY on AMI Originate
ASTERISK-20069: Some events are not logged
ASTERISK-20073: Crash within Say
ASTERISK-20074: TLS stops responding in 1.8.14-rc1 when an aastra phone tries to register
ASTERISK-20076: The command stream_file in Perl-AGI returns weird numbers that nobody pressed
ASTERISK-20078: Dial() command help -- add information about timeout
ASTERISK-20081: Asterisk crashes and restarts randomly when receiving DTMF from a caller
ASTERISK-20082: Phantom calls open analog DAHDI switch and send dialplan into loop, causing /var to fill up with logs
ASTERISK-20085: [patch] this is a test and there is really no patch!
ASTERISK-20086: Realtime queue does not re-read announce variable from mysql after first use - Asterisk Version 1.4
ASTERISK-20087: Asterisk gives "Broken pipe" warning on hangup
ASTERISK-20088: Duplicate ICE candidates and SDP payload truncation
ASTERISK-20090: Crash when running 'core show locks' with BETTER_BACKTRACES enabled
ASTERISK-20092: Invalid queue member state when extension state is ringing & in use.
ASTERISK-20093: sip show field "Def. Username:" displays incorrect value after config file change and asterisk reload while device still registered
ASTERISK-20094: OOH323 doesn't try re-register to GK if registration failed and RRJ is received
ASTERISK-20095: Incorrect Documentation for DEC function
ASTERISK-20097: asterisk disable verbose logging
ASTERISK-20098: P2P bridging can cause the SSRC of a RTP session to change during a call
ASTERISK-20099: NO CDR when ORIGINATE is pointing LOCAL Channel
ASTERISK-20100: CLONE - Asterisk gives "Broken pipe" warning on hangup
ASTERISK-20101: Can't dial from Motif to Google Chat user
ASTERISK-20102: No audio on call from SIP > Motif > Google Chat
ASTERISK-20103: chan_motif causes Asterisk to fail to load with either no motif.conf, or the sample config
ASTERISK-20106: Google Chat with Video > Motif > Echo, No video; or call a H.264 endpoint and no video
ASTERISK-20107: Calling Motif to Google Talk web plug, audio drops after a while
ASTERISK-20109: get_ast_cmd doesn't differentiate between failure and empty list return
ASTERISK-20113: ParkAndAnnounce doesn't return to n+1 when no return_context defined
ASTERISK-20114: Motif doesn't respect codec order preference
ASTERISK-20118: When dnsmgr is enabled, the status of SIP trunks that use host names changes to UNKNOWN after sip reload
ASTERISK-20119: Peer MWI subscriptions erroneously removed when another SIP dialog is destroyed
ASTERISK-20120: Unit test ast_parse_arg_test fails on 32-bit machines
ASTERISK-20122: [Regression] 1.8.14.0 seg faulted 10 times supposedly in ast_ssl_teardown while a TCP TLS node was re-regestering after the Asterisk reload
ASTERISK-20123: Asterisk lists IAX 2 extensions as invalid in queues
ASTERISK-20124: chan_sip retransmit timeout causes a AST_CAUSE_PROTOCOL_ERROR, translates to 603 Declined
ASTERISK-20125: patch for alloca and ast_strdupa checked for return values
ASTERISK-20126: no jump to 's' extension
ASTERISK-20127: [Regression] Config.c config_text_file_load() unescapes semicolons ("\;" -> ";") turning them into comments (corruption) on rewrite of a config file
ASTERISK-20128: Virtualized asterisk.org 1.8.14.0 no longer runs in a KVM virtualized environment. Compiles without error, but fails with Illegal instruction on launch Regression since 1.8.13.0 Last good 1.8.12.2
ASTERISK-20129: Incorrect <CallerID> from 'core show channels verbose' when using Queue()
ASTERISK-20130: Asterisk 1.8.15.0 Blockers
ASTERISK-20131: Asterisk 10.7.0 Blockers
ASTERISK-20132: Security Vulnerability: remote authenticated attacker can execute arbitrary shell commands on system through app ExternalIVR
ASTERISK-20134: app_meetme deprecated in Asterisk 10 should be moved back to extended support
ASTERISK-20135: Use of ast_asprintf and asprintf needs to be checked for failure.
ASTERISK-20136: Asterisk crashes in pjnath when call is teared down quickly after creation
ASTERISK-20140: Segmentation fault on chan_sip
ASTERISK-20145: Segfault in iax_pvt_callid_get
ASTERISK-20148: Confbridge Admin is played hold music when last participant leaves the conference. When a participant returns to the same call the Admin remains hearing hold music.
ASTERISK-20149: Crash when faxing SIP to SIP with strictrtp set to yes
ASTERISK-20150: Segmentation Fault after several rounds of users joining a conference with a "hangup request all"
ASTERISK-20151: No CDR when queue call hangs up while ringing agent and 2 or more agents in that queue are busy
ASTERISK-20156: MixMonitor creates file even if call state is unanswered (leaves zero length files)
ASTERISK-20157: Code Cleanup in app_alarmreceiver
ASTERISK-20158: Add support to Audio Call Next Event - in app_alarmreceiver
ASTERISK-20159: Segfault when starting Confbridge in conf_find_bridge_profile
ASTERISK-20161: asterisk crash on app_queue change mebers status pause/unpause with astdb
ASTERISK-20162: app_queue does no Status update on empty SQL result
ASTERISK-20163: Variables evaluated in dialplan are case insensitive, whereas channel variables/system variables are not
ASTERISK-20167: [patch] UTF-8 cyrillic characters in voicemail email subject cause subject corruption
ASTERISK-20168: audiohook.c: Flushing audiohook
ASTERISK-20169: Channel groups assigned with GROUP() are not cleared after hanging up
ASTERISK-20170: res_odbc crash after freetds dsn reconnects
ASTERISK-20173: segfault Asterisk
ASTERISK-20174: Asterisk becomes unresponsive when tryung to send fax with T38
ASTERISK-20175: Security Vulnerability: denial of service attack through exploitation of device state caching
ASTERISK-20179: Asterisk core sounds, Danish
ASTERISK-20180: answer not found 404 to OPTIONS packet
ASTERISK-20181: Various confbridge features not available when set in user profile within confbridge.conf
ASTERISK-20182: Parsing a label beginning with a numeric character in all Goto/GotoIf/GotoIfTime application causes unexpected behavior
ASTERISK-20183: Intermittent audio
ASTERISK-20184: Asterisk crash in Dial
ASTERISK-20186: Security Vulnerability: IAX2 peer's NEW message bypasses ACL defined in realtime
ASTERISK-20188: Don't stop ChanSpy/Record after call hangup/silence
ASTERISK-20189: add a none rotatestrategy that disables internal log rotation
ASTERISK-20190: WARNING messages spammed :"channel.c: Exceptionally long voice queue length queuing to ..." CPU usage spikes and all extensions enter InUse or OnHold
ASTERISK-20191: syntax error in VERSION script during make
ASTERISK-20192: Despite a "context" defined in queues.conf - queue callers can't exit a queue by dialing extension numbers within that context
ASTERISK-20193: AMI originate produces error "originated failed" when executing executecommand@asterisk_guitools context used by asterisk-gui to make system call.
ASTERISK-20194: SRTP: after hold action no audio on holded peer.
ASTERISK-20197: MeetMe: When using E or e options entering a conference user doesn't hear leading zeros
ASTERISK-20198: Store hangup cause information on the callee channels in addition to the calling channels
ASTERISK-20201: video tos/qos not supported by all asterisk version?
ASTERISK-20202: IP_MTU_DISCOVER incorrect work
ASTERISK-20203: Patch to handle complex SDP from TANDBERG/257 - "Unsupported top-level media type in offer"
ASTERISK-20204: Asterisk not rejecting call setup on CIC that is down
ASTERISK-20206: append new H264 fmtp attr from CISCO Tandberg to res_format_attr_h264
ASTERISK-20207: Asterisk should clear out any .lock files in the voice mail directory on startup.
ASTERISK-20210: chan_sip.c: No compatible codecs for this SIP call
ASTERISK-20211: chan_sip incorrect mapping from sip error codes to internal (ISDN) CAUSECODEs.
ASTERISK-20212: Deadlock / TCP SIP Stack
ASTERISK-20215: Module Support Level - res_curl res_fax
ASTERISK-20216: Bridged channels both have inbound RTP to Asterisk, but no RTP outbound from Asterisk
ASTERISK-20217: AST_EVENT_MWI does not always trigger instant MWI
ASTERISK-20218: Alarmreceiver DTMF strings not recieved correctly from alarm-panel->SPA-PAP2->asterisk
ASTERISK-20219: [patch] - IAX2 Call Encryption Fails with RSA authentication
ASTERISK-20220: ConfBridge not working properly
ASTERISK-20221: seg fault when register via websocket
ASTERISK-20224: Fix Documentation
ASTERISK-20225: Segmentation Fault on manager_play_dtmf sip_senddigit_end
ASTERISK-20226: Segfault in chan_sip while performing connected line update
ASTERISK-20227: Segfault (possible memory corruption?)
ASTERISK-20229: dialing through chan_local breaks t38 fax
ASTERISK-20230: Attended transfer using SIP Refer to a queue fails to play MOH
ASTERISK-20231: codec_ilbc using memcpy instead of memmove for overlapping mem
ASTERISK-20233: SRTP not working with some devices (Eg Grandstream gxv3175) - Message "Can't provide secure audio requested in SDP offer"
ASTERISK-20234: SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
ASTERISK-20236: Certified Asterisk crashing due to TCP/TLS issue.
ASTERISK-20237: Assert failure in res_rtp_asterisk calling pj_mutex_unlock after a few calls
ASTERISK-20238: Registering SIP user from JavaScript sipML5 client fails when random ".invalid" domain sent in Contact header
ASTERISK-20239: JabberSend only accepts recipients that contain '2' instead of '@'
ASTERISK-20240: Incorrect work of __ast_rwlock_timed[rd|wr]lock() functions on OpenBSD
ASTERISK-20241: alignment of MD5Context in buffer not enforced (theoretical issue only)
ASTERISK-20242: Error sending fax: FAX_FAILURE_PROTOCOL_ERROR
ASTERISK-20243: Update documentation for QueueMemberStatus AMI event to reflect actual device state values
ASTERISK-20245: Asterisk manager output for command "Action:queues" is inconsistent from other commands.
ASTERISK-20251: can't delete voicemails
ASTERISK-20253: HangupcauseClear XML Doc issue
ASTERISK-20254: HTTP Error Bindaddr
ASTERISK-20255: Calls to Motif are not working when a video/audio capable client and a chat only client is logged in to gtalk account simultatious
ASTERISK-20256: Motif should set a profile-level-id= in the SDP for incoming video calls
ASTERISK-20257: chan_motif: no audio to Google Talk when SIP phone uses G722
ASTERISK-20258: ODBC default username not root as the comment in res_odbc.conf claims
ASTERISK-20259: [patch] Update Doxygen Configuration for make progdocs
ASTERISK-20260: Increase robustness of ast_tls_cert
ASTERISK-20261: app_conference cause asterisk coredump
ASTERISK-20262: Asterisk console becomes unresponsive under heavy call load
ASTERISK-20263: Document ConfBridge states and marked user, waitmarked user, 'normal' user interactions
ASTERISK-20265: Document PRESENCE_STATE function and CustomProvider device state providers
ASTERISK-20267: Document WebRTC integration with Asterisk
ASTERISK-20269: Document ICE Support in Asterisk
ASTERISK-20271: Document FEATURE/FEATURE_MAP dialplan functions
ASTERISK-20273: Document SIP DirectMedia INVITE glare reduction
ASTERISK-20275: Document Corosync
ASTERISK-20277: Document named pickup groups/callgroups
ASTERISK-20279: Add CODING-GUIDELINES back to SVN or at minimum a link to the wiki
ASTERISK-20280: In app_voicemail we attempt to play the sound "vm-urgent-removed", which should be "vm-marked-nonurgent"
ASTERISK-20281: "core set verbose" behaves strangely, can't alias it, cli.conf example broken
ASTERISK-20282: Call pickup incompatibility with Cisco 1760V + RPID
ASTERISK-20283: Files descriptors for dummy channels need to be set to -1
ASTERISK-20284: The DIAL application does not cause asterisk to start reading packets if we have sent a "Session Progress" and the other end has sent a Re-INVITE.
ASTERISK-20287: Broken hangupcause passtrough
ASTERISK-20288: PhonerLite reports RTP read error when ICE Support Enabled
ASTERISK-20289: [patch] Use ALAW in app_alarmreceiver
ASTERISK-20294: Asterisk is unable to execute AGI commands longer than 2048 bytes
ASTERISK-20295: Asterisk is not incrementing the sequence numbers for the retransmission of the DTMF end packets(RTPEvent packet with end bit set to 1)
ASTERISK-20296: In certain scenarios, asterisk can send rtp in an unsupported payload type to an endpoint
ASTERISK-20297: Asterisk not sending status updates for Custom device hints on RINGING or RINGINUSE states
ASTERISK-20298: Deprecate chan_gtalk, chan_jingle and res_jabber
ASTERISK-20305: Asterisk crashing on Page()
ASTERISK-20308: Asterisk 1.8.16.0 Blockers
ASTERISK-20309: Asterisk 10.8.0 Blockers
ASTERISK-20313: GotoIf redirection to label not working in included extension - document expected behavior on wiki and update sample config
ASTERISK-20315: incomplete SDP when using websocket transport
ASTERISK-20316: If audio file is not exitst then channel is hangup
ASTERISK-20317: DNS long responce freeze chan_sip
ASTERISK-20318: Include channel uniqueid in "AsyncAGI" and "AGIExec" events
ASTERISK-20323: Sporadic timing dependent test failures in one-step-parking due to not waiting for Asterisk to be fully booted
ASTERISK-20325: Comments in configs/func_odbc.conf.sample are not consistent with examples. Missing examples.
ASTERISK-20326: res_odbc blocks datastore while a transaction waits for a lock
ASTERISK-20328: Add new sound prompts to 'extra sounds'
ASTERISK-20331: asterisk should not reinvite on 2 channels with incompatible codecs
ASTERISK-20332: Missing CDR rows on Abandon queue event
ASTERISK-20333: 'directmedia' not working any more
ASTERISK-20335: Crash in ast_cel_report_event
ASTERISK-20337: iax2 provisioning cache mismanaged
ASTERISK-20338: iax2 debug only shows received packets when "iax2 set debug peer xxx" is used
ASTERISK-20339: chan_mgcp, resp_pktccops ast_debug support
ASTERISK-20341: playback of files from asterisk-core-sounds-en-g722-current sound garbled and half speed on Audiocodes and Yealink phones
ASTERISK-20344: RTP/ICE STUN port is not configurable
ASTERISK-20345: Prevent Music on hold from being played when a 183 response indicates the media stream should be sendonly
ASTERISK-20346: Modules need to ensure that any functions, apps, AMI actions, etc. they register are unregistered if the module declines loading
ASTERISK-20347: Asterisk 1.8.16rc1 deadlock in cdr_mysql
ASTERISK-20348: writesql does write to mysql
ASTERISK-20349: DEBUG_MALLOC version of ast_strndup() may cause buffer overflow
ASTERISK-20351: SIP attended transfer - wrong billsec in CDR (transferring side)
ASTERISK-20352: Incorrect handling of SIP 484 Address incomplete
ASTERISK-20353: Wrong dutch date syntax in say.c: function say_date_with_format_nl
ASTERISK-20354: ResetCDR(e) problem - Duration time value is wrong
ASTERISK-20355: [patch] Provide an entry point for stdexten written in AEL from pbx_config.c
ASTERISK-20356: JABBER_RECEIVE timeout in milliseconds, documentation says seconds
ASTERISK-20357: T.38 offer for peer that does not support T.38 fails to warn user on CLI
ASTERISK-20360: XMPP sendtodialplan hangs up after executing first priority (when using Hangup())
ASTERISK-20361: XMPP segfaults
ASTERISK-20362: res_asterisk_rtp: Fix build error when using parallel make
ASTERISK-20366: Build errors on OpenSolaris
ASTERISK-20367: One-way audio with media_address
ASTERISK-20368: res_rtp_asterisk: Eliminate "type-punned pointer" build warning.
ASTERISK-20369: AMI channelvars option can break manager protocol
ASTERISK-20375: Asterisk channel reference leak when attempting to transfer a call originated to a local channel running the Echo application
ASTERISK-20380: Bad ao2_unlock call in app_queue's try_calling
ASTERISK-20381: Asterisk stops registering
ASTERISK-20383: Add missing named call pickup group features for parity with numeric call pickup groups.
ASTERISK-20384: Dialing pickupexten could fail even though there is a call it could have picked up.
ASTERISK-20386: Named call pickup groups implementation improvements.
ASTERISK-20390: chan_local queue members broken by r372050
ASTERISK-20391: No ringback tone on any call from Asterisk to connected NEC PBX Extensions
ASTERISK-20392: OpenSSL headers not picked up when configured using --with-ssl=
ASTERISK-20393: File Structure of the sounds library
ASTERISK-20394: Asterisk crashes in ast_generic_bridge()/ast_frame_free()
ASTERISK-20395: CLI Originate to chan_gtalk causes memory corruption crash
ASTERISK-20396: "manager show commands" output is cropped (privilege column)
ASTERISK-20397: "manager show user <user>" shows the "all" permission despite it not being set
ASTERISK-20398: Asterisk crashing when recording ConfBridge calls (10.7.1)
ASTERISK-20399: Compilation on some systems requires the -fnested-functions flag
ASTERISK-20400: Let ODBC store multiple formats
ASTERISK-20401: [patch] INFO - RFC2833 transcoding, problem in digits regeneration when there is silence
ASTERISK-20402: Unable to cancel (features.conf) attended transfer
ASTERISK-20403: asterisk-extra-sounds LICENSE missing
ASTERISK-20404: sound_only_one gets ignored when there are CONFBRIDGE() settings in the dialplan
ASTERISK-20405: MessageSend() suggests 'from' isn't required in XMPP message responses, but it is
ASTERISK-20406: Make samples sets astsbindir when --prefix is something other that /usr
ASTERISK-20407: Asterisk compilation doesn't set rpath when --prefix is something other that /usr
ASTERISK-20408: constify astobj2's __ao2_ref_debug parameters
ASTERISK-20409: sip_tech_info channels cannot be bridged, not even with themselves
ASTERISK-20410: Asterisk 1.8.17.0 Blockers
ASTERISK-20411: Asterisk 10.9.0 Blockers
ASTERISK-20412: Update Doxygen Configuration for make progdocs
ASTERISK-20414: Timeout antipattern using ast_waitfor_nandfds
ASTERISK-20415: Strict RTP protection learning mode processes non-RTP packets too
ASTERISK-20416: Unknown issue forces maximum calls limit reached at 500 calls - Autodestruct warnings spamming CLI and extension to extension calls not possible
ASTERISK-20417: Nortel transfer problem
ASTERISK-20418: peer Call-ID change after core reload - SIP provider requests that we maintain Call-ID
ASTERISK-20424: Erroneous Multiple DTMF Digit Detection
ASTERISK-20433: Asterisk exceeds allowed stack during RTCP read on openwrt with uclibc / eglibc during feature code blind transfer and LOW_MEMORY option
ASTERISK-20434: Asterisk Core Dumps when reloading res_phoneprov.so
ASTERISK-20435: app_voicemail deletes the wrong greeting if both an unavailable and a temporary greeting is available and imap greetings are used
ASTERISK-20437: Deadlock with ast_context_remove_extension_callerid and handle_request_do
ASTERISK-20439: Do not fail load of chan_sip if res_http_websocket is not loaded
ASTERISK-20440: [patch] No ringback towards SLAstation on outbound trunk call.
ASTERISK-20441: AEL jump to wrong ext after switch-statement is completed (dialplan pattern)
ASTERISK-20442: dtmf callerid regression
ASTERISK-20443: ast_read() on chan '...' called with no recorded file descriptor
ASTERISK-20444: Google to deprecate SASL PLAIN
ASTERISK-20453: res_xmpp.c: JABBER: socket read error, afterwards outgoing connections never get answered
ASTERISK-20454: Can't use SIP realtime registrations alone without realtime peers
ASTERISK-20455: dialplan fails to run the invalid "i" extension due to an uninitialized variable dat_exten in main/pbx.c
ASTERISK-20456: Unable to take calls when MYSQL server is unreachable
ASTERISK-20457: GSM encoding is not thread safe
ASTERISK-20458: ConfBridge() dislplays many ERROR messages on console when loading invalid menu data
ASTERISK-20460: Queue show output is not in order
ASTERISK-20461: channel originate Local/foo * forces translation via slin
ASTERISK-20462: [patch] Trunk not hungup if SLA Station hangs up before answer
ASTERISK-20463: Atxfer manager command (and channel feature) can't transfer to an extension including '#'
ASTERISK-20464: Can't join ConfBridge() with video
ASTERISK-20465: Updates to make work with latest SVN revisions + T.38 support + interoperability improvements + bugfixes
ASTERISK-20466: Clarify specification of the AMI Originate action's Codecs parameter
ASTERISK-20467: Inconsistency in naming convention used with sound files called in app_voicemail
ASTERISK-20469: IMAP Voicemail - asterisk18-voicemail-imapstorage package doesn't install IMAP support in Asterisk
ASTERISK-20474: Segfault Asterisk 1.8.15 on vmware
ASTERISK-20475: Asterisk down because of IAX2 channel is destroyed
ASTERISK-20476: play_file failed for 'queue-seconds'
ASTERISK-20481: Not getting correct Dahdi DIALSTATUS ,& Not Getting CallerID from pstn
ASTERISK-20482: Certain mp3 file will cause crash in format_mp3.c
ASTERISK-20483: Allow Asterisk to report git SHAs in version string.
ASTERISK-20484: Code Cleanup in app_alarmreceiver caused new issue where event are processed before receiving all digits
ASTERISK-20486: MeetMe Unable to write frame to channel after SIP channel hangs up.
ASTERISK-20487: Failure to have OpenSSL w/ SRTP support results in confusing error message
ASTERISK-20492: Stuck DTMF when using ChannelRedirect to split a two channel bridge
ASTERISK-20493: Crash in chan_sip when call initiated from peer with dtlsenable=no sent to peer expecting a DTLS handshake
ASTERISK-20495: Segfault in XMPP caused by the presence stanza of one of my contacts
ASTERISK-20496: Patch to app_voicemail.c for large vmblasts
ASTERISK-20498: Asterisk does not support .invalid IPv6 unspecified address
ASTERISK-20499: Crash in libsrtp srtp_unprotect_rtcp when SIP channel is bridged with non-optimizing Local channel
ASTERISK-20500: ID callerid not update on blind transfer
ASTERISK-20501: Distorted Audio if you Dial multiple numbers at once (DAHDI)
ASTERISK-20502: Definition of new test-condition called sip-channels, which checks for the output of "sip show channels"
ASTERISK-20503: Update Static HTML files
ASTERISK-20504: Asterisk 1.8 should use [compat] section in asterisk.conf and behave acconding to the flag pbx_realtime=1.6
ASTERISK-20505: Migrate hashtest/hashtest2 to be unit tests
ASTERISK-20506: With alwaysauthreject=yes AND allowguest=no Asterisk fails to report Attacker's IP Address
ASTERISK-20507: MWI Refcount Error causing segfault
ASTERISK-20508: Monitor run twice on the same channel. Resulting mix of second Monitor instance has offset or drifting audio legs.
ASTERISK-20509: app_queue parameters setinterfacevar, setqueueentryvar, setqueuevar, membermacro are only used prior to bridging channel, but should happen any time app_queue attempts a connection to the member (regardless of whether it's answered)
ASTERISK-20511: Directrtpsetup does not wrk in SVN-branch-1.8-r374177
ASTERISK-20512: Manager sends a bridge/link message following a hold
ASTERISK-20515: core dump in ast_channel_destructor if chan->tech_pvt is not null
ASTERISK-20520: Write a test for the Asterisk Test Suite that covers directrtpsetup
ASTERISK-20521: Document CDR behavior on the Asterisk wiki
ASTERISK-20524: AMI improperly handles lines of exactly 1025 characters
ASTERISK-20527: AuthID cannot be set for registrations when callbackexten is used
ASTERISK-20528: Handling of natted User Agents behind stateful firewall without SIP support
ASTERISK-20529: Asterisk 10.10.0 Blockers
ASTERISK-20530: Asterisk 1.8.18.0 Blockers
ASTERISK-20531: Asterisk 11.0.0 Blockers
ASTERISK-20532: AGI script fork()s, Asterisk doesn't keep processing the dialplan when parent dies
ASTERISK-20533: Odd happenings in inotify and kqueue code with spool files.
ASTERISK-20537: Asterisk deadlocks between looking up extension from process_sdp and bridge execution from pbx_realtime
ASTERISK-20538: inconsistent ast_verb numbering
ASTERISK-20539: When the peer for an outbound call is defined by a round-robin DNS, the response to the SIP 407 challenge goes to a different server than the original SIP INVITE (and from which the 407 came)
ASTERISK-20541: Implement curses/ncurses terminal width checking and apply it to CLI commands used all over Asterisk
ASTERISK-20542: 'ERROR: getnameinfo(): ai_family not supported' logged when Asterisk rejects a registration
ASTERISK-20544: action_originate called via ast_hook_send_action causes a segfault
ASTERISK-20545: chan_sip loads too early because of exposed global symbols
ASTERISK-20550: Deadlock between SIP pvts being placed in container and CLI command 'sip show channels'
ASTERISK-20551: Segfault when scheduled provisional keepalive is handled - dialog has already been destroyed
ASTERISK-20552: Deadlock when DEBUG_THREADS enabled due to held lock inside of standard locks
ASTERISK-20553: Locks and linked list corruption
ASTERISK-20554: Outgoing calls fail to establish audio due to ICE negotiation failures
ASTERISK-20555: MusicOnHold don't stop the streaming process if not needed
ASTERISK-20556: When the remote peer hangs first asterisk detects a false hangup on FXS: breaks hints and answers subsequent calls without user intervention
ASTERISK-20557: res_calendar_caldav.c:157 SSL handshake failed: SSL error: GnuTLS internal error
ASTERISK-20558: Incorrect TRANSFER entry in queue_log on attended transfer
ASTERISK-20559: SIP TCP/TLS: When checking the CA certificate fails, the call still goes through
ASTERISK-20561: Asterisk 1.8 allows the # character in SIP URI, 10 and higher versions do not - need to document in UPGRADE.txt possibly other places?
ASTERISK-20562: Contact information appears to be mis-parsed and registered incorrectly
ASTERISK-20567: bashism in autosupport
ASTERISK-20569: chan_dahdi does not overwrite context completely.
ASTERISK-20570: Asterisk, when acting as the UAS in Session Timer negotiation, fails to add required header in 200 response ("Require: timer")
ASTERISK-20571: Asterisk crash possibly related to queue announce settings
ASTERISK-20572: Realtime Peers behind NAT are Set to RFC1918 private address after sip reload
ASTERISK-20573: Phone does not re-register after system crash
ASTERISK-20574: Crash in MeetMe using a chan_motif channel when shutting down Asterisk
ASTERISK-20576: compile errors for asterisk 11 - sdl-config: not found
ASTERISK-20577: Asterisk deadlocks waiting for timer in res_timing_pthread while running AGI script
ASTERISK-20578: sip handle_incoming needs more calls to sec. framework
ASTERISK-20579: Asterisk fails to compile on Solaris without makeopts edits
ASTERISK-20580: skypeforasterisk compile error on asterisk-11.0.0-rc2
ASTERISK-20581: segmentation fault with res_fax_digium-10.1_1.3.1-generic_64 on asterisk 11.0.0-rc2
ASTERISK-20589: When using eswitch - variable substitution fails if there is no dialplan executed immediately before the eswitch
ASTERISK-20590: Frequent outgoing gtalk failure resulting in 100% CPU usage
ASTERISK-20591: UDTPL seems not to be loading.
ASTERISK-20592: no sound between chan_motif and psi
ASTERISK-20593: [patch] Future-dated call files are ignored when astspooldir is relative
ASTERISK-20594: Asterisk crash
ASTERISK-20595: After attended transfer asterisk cpu goes to 30% and need to be restarted
ASTERISK-20596: Got SIP response 503 "Max CPS rate exceeded" back from problem!!!
ASTERISK-20597: Cannot forward voice mail due to "Mailbox is full with capacity of 100" even though the folder has less than 100 messages
ASTERISK-20598: Interrupted system call
ASTERISK-20599: [patch] A Module to Submit CDR via Curl to a webservice end point
ASTERISK-20600: Asterisk fails to start
ASTERISK-20601: Confbridge recording does not work
ASTERISK-20603: Crash Asterisk 1.8.1 during SRTP
ASTERISK-20604: CLONE - bad dialog-info remote information
ASTERISK-20605: System ignore some dtmf digits, when some one called by their PBX phones
ASTERISK-20606: Wrong confbridge behavior when participants enter simultaneously
ASTERISK-20610: Asterisk 1.8.13.0 / SpanDSP / ReceiveFAX Fails
ASTERISK-20611: sip registery lost after sip reload
ASTERISK-20612: Segmentation fault related to pthread stack size with LOW MEMORY compiler flag selected
ASTERISK-20613: IAX2 channel fails to transfer
ASTERISK-20614: Investigate and fix sip_attended_transfer_tcp test failures
ASTERISK-20615: Fix sip_outbound_address test failures
ASTERISK-20616: Fix sip_attended_transfer_v6 test failures
ASTERISK-20621: Ajam Demo not working properly
ASTERISK-20622: Default enabling of the "allowguest" setting in Asterisk should be revisited, as it allows systems by default to be potentially vulnerable
ASTERISK-20623: App_queue doesn't increment number of busy agent in certain situations
ASTERISK-20624: add feature to register to sip provider with "authuser@domain"
ASTERISK-20625: asterisk crashed after warning Can't fix up channel
ASTERISK-20626: Add Subscription Context to SIPshowpeer AMI event response
ASTERISK-20627: ConfBridge() crashes if no timing module is loaded
ASTERISK-20628: [patch] - main/pbx.c - ShowDialPlan generates with error if no Exten: was presented and there are no exten => lines present
ASTERISK-20631: Unable to connect via WebRTC
ASTERISK-20633: Asterisk SIP channel handling of MOH media re-INVITES not RFC 3264 compliant
ASTERISK-20634: SIP registrations being lost
ASTERISK-20635: New application to wait on the dialplan until a signal is received
ASTERISK-20637: Can not dial a dahdi channel
ASTERISK-20638: SIP dialog matching is incorrect when multiple provisional responses are received with pedantic SIP checking
ASTERISK-20639: Dynamic hints are not properly initialized when the extension contains an underscore.
ASTERISK-20641: Erroneous error messages from Monitor when using options 'i' and 'o'
ASTERISK-20642: Occasional audio failure after placing call on hold
ASTERISK-20643: SIP ICE support - remove hardcoded limitation on SDP size, make ICE support disabled by default in SIP, maybe provide a better warning message
ASTERISK-20644: Don't always use the existing TCP connection for in-dialog requests
ASTERISK-20645: Outgoing Google Motif Calls connect but continue ringing on outgoing side
ASTERISK-20646: [patch] - manager_shutdown fails to completely shutdown AMI and leaks memory
ASTERISK-20647: [patch] Failure to cleanup SQLite3 statements during exit causes call to sqlite3_close to fail; leaks memory
ASTERISK-20648: [patch] - Memory leaks in xmldoc
ASTERISK-20649: Patches for memory leaks and other cleanup across several files
ASTERISK-20650: Asterisk resets ptime value in 200 OK response
ASTERISK-20652: AppVoicemail creates multiple duplicates of the contents of #included users.conf files.
ASTERISK-20653: Asterisk allows Session-Expires below 90 in a 200 OK
ASTERISK-20654: Core verbosity restricts maximum remote console verbosity
ASTERISK-20655: Cannot reset pin with CONFBRIDGE(user,pin)
ASTERISK-20656: Deadlock in pthread
ASTERISK-20657: saycid in voicemail for french numbers (0123456789)
ASTERISK-20658: Blindly doing alloca (or strdupa) on potentially large user input is bad
ASTERISK-20661: Make voicemail saycid use say.conf for saying caller phone number to customer
ASTERISK-20664: Add new formatting choices for summary output from "sip show peers"
ASTERISK-20670: Wrong error correction in T38 INVITE when multiple Asterisk servers are involved.
ASTERISK-20671: Add Who Hung Up support to the Motif channel driver
ASTERISK-20673: Embeded decode SIP-T/SIP-I
ASTERISK-20674: nat=force_rport,comedia does not behave the same as nat=yes
ASTERISK-20675: [patch] Add return codes to SLATrunk() so that dialplan can identify channel/station that actually answered the call.
ASTERISK-20676: [patch] ${HANGUPCAUSE_KEYS()} returns nothing if I don't Dial()
ASTERISK-20677: Action Challenge not working with allowmultiplelogin=no
ASTERISK-20679: Feature code attended transfer fails silently, each call leg works fine
ASTERISK-20680: Increase the buffer for dynamic feature
ASTERISK-20681: Unable to compile pjproject in Asterisk 11
ASTERISK-20682: deadlock when servicing MWI event
ASTERISK-20683: Ability to execute some macro or gosub on caller's channel
ASTERISK-20684: Installation Issue
ASTERISK-20685: ABE does not respond with ACK on retransmission of 200 OK after it sent ACK
ASTERISK-20686: Confbridge module not loading successfully after changing
ASTERISK-20687: make menuselect fails on asterisk 1.8 due to unreferenced linker lib -ltinfo
ASTERISK-20688: IAX2 authdebug=no suppresses Logger/CLI Notice messages of failed authorization attempts (invalid user, bad password)
ASTERISK-20689: Sporadic failure in Masquerade test due to parking timeout
ASTERISK-20690: Call gets stuck in Queue and all subsequent calls to the queue do not get delivered to available agents.
ASTERISK-20691: Create a generic thread pool for Asterisk
ASTERISK-20692: Implement a Data Access Layer for the new SIP channel driver
ASTERISK-20693: Data Access Layer - Phase One - Definition of Objects
ASTERISK-20696: Wideband/variable bandwidth modification of app_jack
ASTERISK-20701: Siren14 does not translate from/to slin32
ASTERISK-20702: res_rtp_asterisk.c:3455 ast_rtp_read: RTP Read error: Invalid or incomplete multibyte or wide character. Hanging up.
ASTERISK-20703: Merge menuselect into the Asterisk source
ASTERISK-20705: Call Recording on Inbound Calls causes Asterisk 10.10.0 to crash (Segfault)
ASTERISK-20706: reduce CLI spamming of "Extension Changed" device state messages
ASTERISK-20708: Deadlock in chan_sip on transfer when trying to update redirecting information
ASTERISK-20709: Asterisk sends wrong Caller IDs in first SIP NOTIFY message
ASTERISK-20710: Queue_log all calls ended like COMPLETECALLER
ASTERISK-20711: Wrapup time not working
ASTERISK-20713: Data Access Layer - Phase Two - Initial In Memory Object Implementation
ASTERISK-20715: REGEX function ignores shorthand character starting with backslash
ASTERISK-20716: "s" extension in comebackcontext not honored
ASTERISK-20717: Voicemail access "SQL Get Data error! coltitle=msg_id"
ASTERISK-20718: Asterisk crashes or locks on 'queue show'
ASTERISK-20719: Monitor recordings change filename unexpectedly after a blind transfer, but not attended transfer
ASTERISK-20720: [patch]Accept MWI subscriptions with several elements in Accept header
ASTERISK-20721: fake DTMF
ASTERISK-20722: Preventing Password attacks
ASTERISK-20723: Asterisk warns about pipe characters in string variable created using Set
ASTERISK-20724: Fix natdetected flag being set when VIA doesn't include port in address
ASTERISK-20725: Asterisk API stabilization - Channel Handle/UUID
ASTERISK-20726: Add UUID support to Asterisk
ASTERISK-20727: Certified Asterisk - while DPMA module is loaded - Core Reload on CLI triggers crash
ASTERISK-20728: Add UUID handle to channel struct
ASTERISK-20729: Document UUID handle for channels
ASTERISK-20730: Add unit tests for UUID library/channel creation
ASTERISK-20736: Add the ability to query for channels based on their handle value
ASTERISK-20742: [patch] Log callers entering/leaving MOH.
ASTERISK-20743: Queue Log - All Calls End With COMPLETECALLER When h Extension Is Present
ASTERISK-20744: [patch] Security event logging does not work over syslog
ASTERISK-20745: In MESSAGE received over WebSocket, the body last char is cut
ASTERISK-20747: [patch] SLA outbound calls do not record accurate CDR record.
ASTERISK-20748: Asterisk system didn't accept some dtmf digits when It called by a external PBX phone
ASTERISK-20749: Cannot enable res_timing_kqueue via "make menuselect"
ASTERISK-20750: res_timing_kqueue makes Asterisk use 100% CPU
ASTERISK-20751: chan_motif leaves UDP ports open
ASTERISK-20752: [patch] - ringing/progress on branched calls not working correctly on some branched calls
ASTERISK-20754: rtp_engine's RTCP{Sent,Received} events should contain the Channel name
ASTERISK-20755: Asterisk 11.0.1 Returns 484 Address Incomplete when sending Messages
ASTERISK-20756: Asterisk sippeers.sql columns place error cause peer to be without codecs when setting disallow=all under MySQL
ASTERISK-20757: Deadlock with 1.8 certified when abandoning queue calls
ASTERISK-20758: No audio when P2P bridge occurs
ASTERISK-20759: Ability to cause serious damage to your asterisk by manipulating sip2cause tables
ASTERISK-20761: Asterisk 11 Compile errors when embedding modules (/usr/bin/ld: Dwarf Error: ...)
ASTERISK-20762: Asterisk Crash, assertion failed, in res_rtp_asterisk thread (ice_worker_thread)
ASTERISK-20763: Memory Leak in chan_sip with TLS enabled clients
ASTERISK-20766: Open Blockers for 1.8.19.0
ASTERISK-20767: Open Blockers for 10.11.0
ASTERISK-20769: Memory leak of local_pvt in chan_local.
ASTERISK-20770: Remote console verbosity implementation causes problems
ASTERISK-20772: Loop bug in ast_rtp_lookup_mime_multiple2() [main/rtp_engine.c]
ASTERISK-20773: fake DTMF or talkoff
ASTERISK-20776: Confbridge code understanding and maintenance.
ASTERISK-20778: Data Access Layer - Phase Three - Implement Generic DAL
ASTERISK-20779: asterisk dailplan Extension locked after 8 o'clock pm and unlock automatically 8 o'clock am
ASTERISK-20780: Asterisk responds to SIP CANCEL with 481 Call/Transaction Does Not Exist
ASTERISK-20781: [patch] One leg of the call is locked when a session timer below 60 is received by the asterisk.
ASTERISK-20782: Allow SayAlpha to announce "Uppercase <letter>" in a string.
ASTERISK-20783: segfault on hangup
ASTERISK-20784: Failure to receive an ACK to a SIP Re-INVITE results in a SIP channel leak
ASTERISK-20785: Cannot delete voicemail at random
ASTERISK-20786: Taskprocessors can leak memory
ASTERISK-20787: Asterisk should inspect Min-SE header in an INVITE even if there is no Session-Expires present
ASTERISK-20788: Add G722 support for chan_skinny
ASTERISK-20789: Make skinny debug tab completion helpful
ASTERISK-20790: skinny does not respect globally set vmexten
ASTERISK-20791: Asterisk Message outside of call stops working
ASTERISK-20792: Segfault during calloc, core dump shows logging string at requested pointer address
ASTERISK-20800: 'module reload app_playback.so' won't load say.conf if it didn't exist during module's first load
ASTERISK-20801: Non-SIP queue members get no calls when ringinuse=no.
ASTERISK-20805: SIP Notify message has incorrect IP address in FROM field
ASTERISK-20806: Strange values reported by CHANNEL(rtpqos,audio,all) - outbound call
ASTERISK-20807: 1.8.20.0 Blockers
ASTERISK-20808: 10.12.0 Blockers
ASTERISK-20809: 11.2.0 Blockers
ASTERISK-20810: Create res_sip
ASTERISK-20811: Create a packageable pjproject
ASTERISK-20813: Create a Git repo for pjproject
ASTERISK-20814: Fix all build warnings in pjproject
ASTERISK-20815: Fix pjproject's build system to be tolerant of build errors and parallel building
ASTERISK-20816: Create a shared object target for PJLIBUtil (and friends)
ASTERISK-20817: Create a shared object target for PJLIB
ASTERISK-20818: Create a shared object target for PJMedia
ASTERISK-20819: Create a shared object target for PJNATHelper
ASTERISK-20820: Create a shared object target for PJSIP
ASTERISK-20821: Create a shared object target for PJSIPUA
ASTERISK-20822: Create a shared object target for PJSIPSIMPLE
ASTERISK-20823: Update repotool scripts to produce a pjproject tarball
ASTERISK-20824: Create a CentOS package from the git repo
ASTERISK-20825: Update Asterisk trunk to look for pjproject as a dependency
ASTERISK-20826: Replace last few tabs with spaces in causes.h
ASTERISK-20827: AMI events for ConfBridge Mute,Record, start and stop
ASTERISK-20829: Crash during normal operation
ASTERISK-20830: Asterisk crash
ASTERISK-20831: [patch] Restarted Asterisk process remains in original directory despite symlink change
ASTERISK-20833: Didn't get a frame from channel: Agent/1127 when hitting the star (*) key when connected to a call in queue.
ASTERISK-20834: Call Recording Fails (mixmonitor)
ASTERISK-20835: RTP not modified when UAS responds with an OK(200) with other ptime then 20ms
ASTERISK-20837: [patch] build_route fails to parse Record-Route headers longer than 255 characters
ASTERISK-20838: Crash - segfault at 13bc ip 00007f7ecb492911 sp 00007f7e817ed9e8 error 6 in libc-2.15.so
ASTERISK-20839: Avoid processing hold control frame when the peer is already in hold
ASTERISK-20840: Speex bitrate won't change
ASTERISK-20841: fromdomain not honored on outbound INVITE request
ASTERISK-20842: Add Queue Pause Device States
ASTERISK-20843: iLBC speech cracks on IAX2
ASTERISK-20848: [patch] ReadExten timeout interrupts audio file playback
ASTERISK-20849: SDP crypto attribute is not well formed in the SDP ANSWER
ASTERISK-20850: [patch]Nested functions aren't portable. Adapting RAII_VAR to use clang/llvm blocks to get the same/similar functionality.
ASTERISK-20851: Asterisk user/group flags ignored by daemon function in /etc/rc.d/init.d/functions
ASTERISK-20852: asterisk/strings.h: struct ast_str used before its declaration
ASTERISK-20853: compile error chan_skype 1.1.4 with asterisk >= rev. 378320 (asterisk 10) & >= rev. 378303 (asterisk 1.8)
ASTERISK-20854: app_minivm core dump in ast_str_encode_mime
ASTERISK-20855: 'sip reload' does not properly flush MWI outbound subscriptions defined in sip.conf
ASTERISK-20856: Segmentation fault in res_rtp_asterisk.so caused by NULL data pointer in frame from sig_analog
ASTERISK-20858: app_minivm fails to clean up mkstemp files
ASTERISK-20860: Create chan_gulp - Phase 1
ASTERISK-20862: Asterisk min and max member penalties not honored when set with 0
ASTERISK-20865: Create a static configuration sorcery wizard using a new schema for chan_gulp
ASTERISK-20867: Create a realtime configuration sorcery wizard using a new schema for chan_gulp
ASTERISK-20869: Bridge API Enhancements - add support for native bridging
ASTERISK-20871: Bridge API Enhancements - add timed feature for breaking a bridge
ASTERISK-20872: Bridge API Enhancements - add timed feature for playback of messages
ASTERISK-20873: Bridge API Enhancements - Refactor Bridge application/AMI action to use Bridging API
ASTERISK-20874: Bridge API Enhancements - add control frame support/processing
ASTERISK-20880: Add channel caching to event subsystem
ASTERISK-20882: Make AsyncAGI actually asynchronous; support asynchronous media operations
ASTERISK-20886: [patch] LDAP configuration and documentation updates
ASTERISK-20887: Add RESTful HTTP interface for Asterisk
ASTERISK-20888: Add JSON API to Asterisk
ASTERISK-20890: Add RESTful framework to Asterisk http server
ASTERISK-20891: Flesh out RESTful API's
ASTERISK-20897: case sensitive match against T.38 params causes T38MaxBitRate to be negotiated at 2400 baud instead of 14400
ASTERISK-20898: sound_only_one parameter will be ignored in confbridge.conf
ASTERISK-20899: Dundi Lookup for Other Channel Drivers returng OTHER/[Mapping]
ASTERISK-20900: cham_motif fails to load
ASTERISK-20901: Security Vulnerability: Possible stack corruption in when parsing H.264 format attributes
ASTERISK-20902: Asterisk waits for AGI to complete, even though we've daemonized it
ASTERISK-20903: Casing error in AMI events for Confbridge
ASTERISK-20904: RFC1918 NAT Issue On Prune
ASTERISK-20905: Asterisk 200OK offers RTP/AVP for video when it should be RTP/SAVP due to SRTP (encryption=yes) being enabled
ASTERISK-20906: DTMF in SIP not working after HOLD / UNHOLD
ASTERISK-20908: Asterisk presents media desc for video in SDP, missing terminating CRLF
ASTERISK-20913: On install, extra sounds for codec g722 fail
ASTERISK-20914: Segfault when iLBC voice frame is interpolated in a jitter buffer due to codec_ilbc's improper manipulation of datalen
ASTERISK-20916: GoogleVoice calls don't connect, but continue ringing despite call having been answered
ASTERISK-20918: Asterisk fails to CANCEL all calls that it initiated via AGI
ASTERISK-20919: Unable to send or receive faxes with Asterisk 11 using res_fax and spandsp-0.0.6 (Tar ball spandsp-20120415)
ASTERISK-20921: CentOS 6.x RPM Support
ASTERISK-20928: DTMF Not Recognised
ASTERISK-20929: Core-dump on SIP BYE for an invalid call transaction
ASTERISK-20930: Wrong password for google voice eventually causes asterisk to crash
ASTERISK-20932: Don't work IP Key Expansion Module for Nortel IP Phone 2004
ASTERISK-20933: No CDR created after call has been split and then bridged back
ASTERISK-20934: Crash in res_srtp.so when SIP channel is bridged with non-optimizing Local channel
ASTERISK-20938: [patch] ConfBridge list from CLI and Manager no longer include waiting members
ASTERISK-20939: Not working parameter mailcmd in voicemail.conf unless Asterisk is running as root
ASTERISK-20945: "Unable to connect to remote asterisk" message on service asterisk start, even though service is running
ASTERISK-20947: astcanary exits immediately because of wrong pid argument
ASTERISK-20949: Core dump on ReceiveFax
ASTERISK-20950: Add configuration support to the new SIP channel driver for the 'standard' transports
ASTERISK-20952: Create pjproject transport module for WebSockets for SIP over WS support
ASTERISK-20953: Create default authentication provider for chan_gulp
ASTERISK-20954: Crash Segmentation Fault in ast_timer_enable_continuous
ASTERISK-20955: Create default endpoint identification services for the new SIP channel driver
ASTERISK-20959: Create Stasis Core Module
ASTERISK-20960: Make it so chan_gulp can actually place a call
ASTERISK-20962: CLONE - "Unable to connect to remote asterisk" message on service asterisk start, even though service is running
ASTERISK-20963: Document that chan_iax2 requires a timing source post 1.8.19.0; ensure a timing source is loaded
ASTERISK-20964: Device call logging has issues.
ASTERISK-20965: Add variable length callinfo packets
ASTERISK-20966: Getting UDPTL(SIP):Transmission error:Resource temporarily unavailable
ASTERISK-20967: Security Vulnerability: DoS attack possible due to fix for CVE-2012-5976
ASTERISK-20969: Fix func_channel documentation for sip/iax2/dadhi
ASTERISK-20970: ignore wrap up time in attended transfer call in queue
ASTERISK-20971: Asterisk hang occasionally
ASTERISK-20972: Race condition between hangup and invite processing
ASTERISK-20973: Insecure=very does not work if callerId found in extention
ASTERISK-20974: chan_iax2 messing with stale channels
ASTERISK-20975: DTMF issue with SIP trunk
ASTERISK-20976: Asterisk 11.2.1 RTP packetization setting ignored in RTP output
ASTERISK-20977: In Asterisk 11.2.1 (verified that this also happens in 10.9.0) if you run ./configure with --disable-xmldoc, core show application xxx will produce an empty output
ASTERISK-20978: Verbose logs don't display after Asterisk rebuild
ASTERISK-20979: Asterisk fails to start, crashes with Assertion error
ASTERISK-20980: [patch] ./configure fails with ptlib 2.10.9
ASTERISK-20981: Call-ID header generation to exclude colon
ASTERISK-20982: xmpp Segfault when delete node using cli
ASTERISK-20983: 'xmpp list nodes' returns (null) node names when collection is defined
ASTERISK-20984: Audible clicks when playing sox encoded au file with STREAM FILE AGI command
ASTERISK-20986: QUEUE_MEMBER 's description is inaccurate
ASTERISK-20987: non-admin users, who join muted conference are not being muted
ASTERISK-20988: Handling of fragmented skinny packets
ASTERISK-20990: Confbridge announcement not played
ASTERISK-20991: Confbridge errors on leaving
ASTERISK-20994: AMI command reception after app_confbridge.so unload results in crash